Introduction

The Aastra 480i looks similar to its sister ADSI model (480e), but has 4 extra buttons for virtual lines. Supports business class SIP. Works well with Asterisk.

The phone is manufactured by Aastra. Firmware 1.2.5 and earlier was developed by Sayson, Firmware 1.3 and later is developed and supported by Aastra. These phones are available with various brandings, including Aastra, Sayson and other VAR's.

In addition an RPM is available for trixbox. If you are using trixbox 2.0 beta. You will see aastra-iphone package on the list of available packages. ( once you install it, it actually disappears from the list, Thats just a trixbox 2.0 beta bug and should be fixed by the final release. The firmware correctly installs into the tftpboot directory) you can also use the RPM with trixbox 1.2.3 by using yum.

Limitations

Note: Please do not remove any of these limitations unless you specifically address the issue you are removing!

Basic phone behavior

MAJOR Using firmware 1.4.1 (and a 1.4.2 beta), the Asterisk application Congestion() (using Asterisk 1.2.x) will frequently cause the phone to lockup. Changing the dialplan to use Playtones(congestion) before hanging up is a workaround. This lockup is also seen in the 9133i.

MAJOR If you go offhook to dial and a call comes in at the same time, your dial sequence will be reset and you can't dial anything for as long as the phone in ringing (imagine this with ringing groups, this sometimes results in the phone answering the call without ringing). The phone should block incoming call ringing if it is currently in the interactive dialing state. In release 1.4.2, if you go offhook to dial and a call comes in at the same time, the incoming call still takes over, but now you can press IGNORE to cancel the new incoming call and resume dialing your outgoing call. However, there is still no way to configure the phone to block the incoming call and be able to dial an outgoing call without interruption.

MAJOR Phones freeze up periodically in firmware releases 1.3.x and 1.4.0. Same issue has been reported for 480i CT, 480i, 9133i, 9122i. Nightly reboots minimize the frequency in which this occurs. (kheston 2006-10-29)

When in do-not-distrub mode, the indicator (a very small icon on the screen) is very subtle and easy to miss (subjective, based on personal preference)

MINOR SIP NAT PORT directive does very little- it changes the contact header but does not actually change the port that the phone uses, making this option near-useless for putting many phones behind the same NAT. This has been confirmed as a bug by AAstra support, fix may be coming in 1.4.2.

By default, you can't dial any number with an asterisk in it. For example, we have "9" as the prefix for the PSTN, and we need to use "*99" to access telco voicemail. If I dial "9*99" (which works with other phones) I get an immediate dial tone. No request shows up at the SIP proxy server. This is easily fixed by modifying the default dial plan, but does require a server from which to download a configuration file.

If you are talking on L1 (while nothing else is going on) and press "Goodbye", the phone ends the current call on L1. This is very handy if you use a headset. In release 1.4.1, if you are talking on L1 when a new call comes in on L2, if you press "Goodbye" to end the current call on L1, the phone will cancel the new call on L2 instead. In release 1.4.2, you can configure the phone so that pressing "Goodbye" will end the current call on L1 and give you the opportunity to answer the new call on L2. If, however, the person that you're talking to on L1 happens to end the call before you get the chance to press "Goodbye", since there is no longer an active call, then the "Goodbye" button will still cancel the new call on L2. If you use a headset and often juggle more than one phone call, it is very easy to cancel incoming calls by mistake.

SIP Issues

If using callerid="Unknown Caller" or anything with a space in it in the general settings under sip.conf, the phone will not "ring". (this is done to replace the default "asterisk"). If no CID info is available. Asterisk will Dial the SIP channel as normal, using the spaced entry (ex: Unknown Caller) but the phone does not ring. (tested with 480i ct firmware 1.4.2.3000)

Limited codec preference control

There is no support for STUN

The phone displays the SIP To: caller information on the called party's screen for some reason.

In the configuration of the phone, ensure that the sip proxy is specified...even if it is the same as the registrar!

Other

There is little documentation on how to specify a dial-plan. See the Dial Plan Configuration page for what there is.

Downloads (Firmware, Docs and other files)

Primary location: Contact and work through the dealer who sold you your phones. They often conduct testing to verify the new firmware. They may also have value added documentation more appropriate for your use then the docs from the manufacturer.

Note: You should avoid downloading these files from an unknown source. This is the firmware of your phone and it is not always possible to restore your phone without sending it in if things get too messed up.

Updating Firmware tftp issues?

While trying to update an early version of the firmware it would not work with a unix tftp server. The solution was to use a windows-based tftp server.

The phone is very sensitive to the TFTP timeout on the server. Try increasing the timeout to 2 or more seconds on the TFTP server. For example on Fedora Core add '-T 5000000' to the server_args line in /etc/xinetd.d/tftp file to increase the timeout to 5 seconds

Introduction

The Aastra 480i looks similar to its sister ADSI model (480e), but has 4 extra buttons for virtual lines. Supports business class SIP. Works well with Asterisk.

The phone is manufactured by Aastra. Firmware 1.2.5 and earlier was developed by Sayson, Firmware 1.3 and later is developed and supported by Aastra. These phones are available with various brandings, including Aastra, Sayson and other VAR's.

In addition an RPM is available for trixbox. If you are using trixbox 2.0 beta. You will see aastra-iphone package on the list of available packages. ( once you install it, it actually disappears from the list, Thats just a trixbox 2.0 beta bug and should be fixed by the final release. The firmware correctly installs into the tftpboot directory) you can also use the RPM with trixbox 1.2.3 by using yum.

Limitations

Note: Please do not remove any of these limitations unless you specifically address the issue you are removing!

Basic phone behavior

MAJOR Using firmware 1.4.1 (and a 1.4.2 beta), the Asterisk application Congestion() (using Asterisk 1.2.x) will frequently cause the phone to lockup. Changing the dialplan to use Playtones(congestion) before hanging up is a workaround. This lockup is also seen in the 9133i.

MAJOR If you go offhook to dial and a call comes in at the same time, your dial sequence will be reset and you can't dial anything for as long as the phone in ringing (imagine this with ringing groups, this sometimes results in the phone answering the call without ringing). The phone should block incoming call ringing if it is currently in the interactive dialing state. In release 1.4.2, if you go offhook to dial and a call comes in at the same time, the incoming call still takes over, but now you can press IGNORE to cancel the new incoming call and resume dialing your outgoing call. However, there is still no way to configure the phone to block the incoming call and be able to dial an outgoing call without interruption.

MAJOR Phones freeze up periodically in firmware releases 1.3.x and 1.4.0. Same issue has been reported for 480i CT, 480i, 9133i, 9122i. Nightly reboots minimize the frequency in which this occurs. (kheston 2006-10-29)

When in do-not-distrub mode, the indicator (a very small icon on the screen) is very subtle and easy to miss (subjective, based on personal preference)

MINOR SIP NAT PORT directive does very little- it changes the contact header but does not actually change the port that the phone uses, making this option near-useless for putting many phones behind the same NAT. This has been confirmed as a bug by AAstra support, fix may be coming in 1.4.2.

By default, you can't dial any number with an asterisk in it. For example, we have "9" as the prefix for the PSTN, and we need to use "*99" to access telco voicemail. If I dial "9*99" (which works with other phones) I get an immediate dial tone. No request shows up at the SIP proxy server. This is easily fixed by modifying the default dial plan, but does require a server from which to download a configuration file.

If you are talking on L1 (while nothing else is going on) and press "Goodbye", the phone ends the current call on L1. This is very handy if you use a headset. In release 1.4.1, if you are talking on L1 when a new call comes in on L2, if you press "Goodbye" to end the current call on L1, the phone will cancel the new call on L2 instead. In release 1.4.2, you can configure the phone so that pressing "Goodbye" will end the current call on L1 and give you the opportunity to answer the new call on L2. If, however, the person that you're talking to on L1 happens to end the call before you get the chance to press "Goodbye", since there is no longer an active call, then the "Goodbye" button will still cancel the new call on L2. If you use a headset and often juggle more than one phone call, it is very easy to cancel incoming calls by mistake.

SIP Issues

If using callerid="Unknown Caller" or anything with a space in it in the general settings under sip.conf, the phone will not "ring". (this is done to replace the default "asterisk"). If no CID info is available. Asterisk will Dial the SIP channel as normal, using the spaced entry (ex: Unknown Caller) but the phone does not ring. (tested with 480i ct firmware 1.4.2.3000)

Limited codec preference control

There is no support for STUN

The phone displays the SIP To: caller information on the called party's screen for some reason.

In the configuration of the phone, ensure that the sip proxy is specified...even if it is the same as the registrar!

Other

There is little documentation on how to specify a dial-plan. See the Dial Plan Configuration page for what there is.

Downloads (Firmware, Docs and other files)

Primary location: Contact and work through the dealer who sold you your phones. They often conduct testing to verify the new firmware. They may also have value added documentation more appropriate for your use then the docs from the manufacturer.

Note: You should avoid downloading these files from an unknown source. This is the firmware of your phone and it is not always possible to restore your phone without sending it in if things get too messed up.

Updating Firmware tftp issues?

While trying to update an early version of the firmware it would not work with a unix tftp server. The solution was to use a windows-based tftp server.

The phone is very sensitive to the TFTP timeout on the server. Try increasing the timeout to 2 or more seconds on the TFTP server. For example on Fedora Core add '-T 5000000' to the server_args line in /etc/xinetd.d/tftp file to increase the timeout to 5 seconds